Switch to using new ACM methods for encoder management

webrtc/voice_engine/channel.h

Index: webrtc/voice_engine/channel.h
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 07a0f789da8f22c907658f8e0ca838f8c7d975e5..c66585311c877b1f227c220eae3876c6a2f66d85 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -17,6 +17,8 @@
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
@@ -480,6 +482,8 @@ class Channel
   TelephoneEventHandler* telephone_event_handler_;
   std::unique_ptr<RtpRtcp> _rtpRtcpModule;
   std::unique_ptr<AudioCodingModule> audio_coding_;
+  acm2::CodecManager codec_manager_;
+  acm2::RentACodec rent_a_codec_;
   std::unique_ptr<AudioSinkInterface> audio_sink_;
   AudioLevel _outputAudioLevel;
   bool _externalTransport;